[asterisk-bugs] [JIRA] (ASTERISK-20780) Sip trunk - 481 Call/Transaction Does Not Exist

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-20780?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-20780:
------------------------------------

    Assignee: mitja
      Status: Waiting for Feedback  (was: Triage)

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

We need additional information to look into this issue.

Please provide the extensions.conf code or dialplan in use for the inbound call and a full log captured during the entire call that demonstrates the issue. https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
Be sure that VERBOSE and DEBUG are set to at least level 5. 

We'll need a SIP packet capture (tcpdump or wireshark) of the same call, so that we can correlate it along with what is in the full log.

Please attach the files to the issue using 'More Actions > Attach Files'

                
> Sip trunk - 481 Call/Transaction Does Not Exist
> -----------------------------------------------
>
>                 Key: ASTERISK-20780
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20780
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 1.8.15.0
>            Reporter: mitja
>            Assignee: mitja
>         Attachments: Scenario1.txt, Scenario2.txt
>
>
> Hi. 
> First scenario:
> I was testing sip trunk on asterisk 1.8, when incoming call is initiated and outside user then hangup without answering, phone will not stop ringing. The provider send me a Cancel request (Request-Line: CANCEL sip:77XXXXXX at 192.168.0.11:5060 SIP/2.0) but then asterisk give me " Status-Line: SIP/2.0 481 Call/Transaction Does Not Exist" error. In these case we have to restart asterisk to free a channel. In earlier versions of asterisk  (1.4,1.6) with same settins these problem never occurred, but only in asterisk 1.8.
> Second scenario:
> When incoming call is initiated and enduser answers the call, than all other calls hangup normaly, until asterisk is reloaded or sip trunk is reregistered.
> I searched the internet for a few days now but I did not found nothing, I did try some patches for reinvite issues but nothing worked.
> Thanks for your answer
> bye, Mitja

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